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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_utility.h

Issue 1802993002: Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_7
Patch Set: Reanimate CreateAudioReceiver() with 5 params, to not break downstream code. Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
13 13
14 #include <map> 14 #include <map>
15 15
16 #include "webrtc/base/deprecation.h" 16 #include "webrtc/base/deprecation.h"
17 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" 17 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
20 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
21 #include "webrtc/typedefs.h" 21 #include "webrtc/typedefs.h"
22 22
23 namespace webrtc { 23 namespace webrtc {
24 24
25 const uint8_t kRtpMarkerBitMask = 0x80; 25 const uint8_t kRtpMarkerBitMask = 0x80;
26 26
27 RtpData* NullObjectRtpData(); 27 RtpData* NullObjectRtpData();
28 RtpFeedback* NullObjectRtpFeedback(); 28 RtpFeedback* NullObjectRtpFeedback();
29 RtpAudioFeedback* NullObjectRtpAudioFeedback();
30 ReceiveStatistics* NullObjectReceiveStatistics(); 29 ReceiveStatistics* NullObjectReceiveStatistics();
31 30
32 namespace RtpUtility { 31 namespace RtpUtility {
33 32
34 struct Payload { 33 struct Payload {
35 char name[RTP_PAYLOAD_NAME_SIZE]; 34 char name[RTP_PAYLOAD_NAME_SIZE];
36 bool audio; 35 bool audio;
37 PayloadUnion typeSpecific; 36 PayloadUnion typeSpecific;
38 }; 37 };
39 38
(...skipping 28 matching lines...) Expand all
68 uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd, 67 uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
69 const uint8_t* ptr) const; 68 const uint8_t* ptr) const;
70 69
71 const uint8_t* const _ptrRTPDataBegin; 70 const uint8_t* const _ptrRTPDataBegin;
72 const uint8_t* const _ptrRTPDataEnd; 71 const uint8_t* const _ptrRTPDataEnd;
73 }; 72 };
74 } // namespace RtpUtility 73 } // namespace RtpUtility
75 } // namespace webrtc 74 } // namespace webrtc
76 75
77 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_ 76 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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